And join when ready

// we have to wait until it's ready
webrtc.on('readyToCall', function () {
// you can name it anything
webrtc.joinRoom('your awesome room name');
});

Need More Control?

In order to make WebRTC as approachable as possible, SimpleWebRTC assumes a lot about the type of app you want to build.

It's unlikely that if you're going to ship an app that uses WebRTC SimpleWebRTC will have the exact features you want. Likely you'll want to use bits and pieces of it. Well, you're in luck! SimpleWebRTC is actually comprised of a whole bunch of independent little modules to help you:

More Info

Henrik's also written a book about modern clientside apps to help you build a well-structured, maintainable app to contain your WebRTC experience. The approaches it describes were used to build Talky and many similar apps. Check out the book: Human JavaScript.

Signaling server

SimpleWebRTC uses the SimpleWebRTC.com sandbox server and is only for development and testing purposes. This server does not provide media relay facilities so there might be connectivity issues.

The signaling server is open source (MIT) licensed as well. You can find it here: github.com/andyet/signalmaster. For production use, please run your own signaling server.

STUN and TURN server

STUN and TURN servers are required to help establish the connection between clients. STUN helps clients to determine their public IP and TURN provides media relay.